WO2006075563A1 - Dispositif de codage audio, methode de codage audio et programme de codage audio - Google Patents

Dispositif de codage audio, methode de codage audio et programme de codage audio Download PDF

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Publication number
WO2006075563A1
WO2006075563A1 PCT/JP2006/300112 JP2006300112W WO2006075563A1 WO 2006075563 A1 WO2006075563 A1 WO 2006075563A1 JP 2006300112 W JP2006300112 W JP 2006300112W WO 2006075563 A1 WO2006075563 A1 WO 2006075563A1
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Prior art keywords
high frequency
noise level
frequency component
signal
correction coefficient
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PCT/JP2006/300112
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English (en)
Japanese (ja)
Inventor
Osamu Shimada
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Nec Corporation
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Priority to EP06702057.8A priority Critical patent/EP1840874B1/fr
Priority to JP2006552903A priority patent/JP5224017B2/ja
Publication of WO2006075563A1 publication Critical patent/WO2006075563A1/fr

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/20—Vocoders using multiple modes using sound class specific coding, hybrid encoders or object based coding
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
    • G10L19/0208—Subband vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques

Definitions

  • Audio encoding apparatus Audio encoding apparatus, audio encoding method, and audio encoding program
  • the present invention relates to an audio encoding device, an audio encoding method, and an audio encoding program, and in particular, an audio encoding device and an audio encoding method for encoding a wideband audio signal with a small amount of information and high quality. And audio encoding program.
  • SBR is intended to compensate for high frequency band signals (high frequency components) that are lost by audio code processing such as AAC, or band-limiting processing based on it, and is compensated by SBR.
  • the information encoded by the SBR includes information for generating a pseudo high-frequency component based on the low-frequency component transmitted using other means. By adding a pseudo high-frequency component to, the sound quality degradation due to band limitation is compensated.
  • FIG. 6 is a diagram illustrating an example of a band extension coding / decoding device using SBR.
  • the encoding side is composed of an input signal dividing unit 100, a low frequency component encoding unit 101, a high frequency component encoding unit 102, and a bit stream multiplexing unit 103
  • the decoding side is a bit stream separating unit 200, a low frequency component A component decoding unit 201, a subband division unit 202, a band extension unit 203, and a subband synthesis unit 204 are configured.
  • the input signal dividing unit 100 analyzes the input signal 1000 and outputs a high frequency sub-band signal 1001 divided into a plurality of high frequency bands and a low frequency signal 1002 including a low frequency component. .
  • the low-frequency signal 1002 is encoded into the low-frequency component information 1004 by the low-frequency component encoding unit 101 using the above-described encoding method such as AAC, and transmitted to the bit stream multiplexing unit 103.
  • the high frequency component encoding unit 102 extracts the high frequency energy information 1102 and the additional signal information 1103 from the high frequency subband signal 1001 and transmits them to the bit stream multiplexing unit 103.
  • the bitstream multiplexing unit 103 multiplexes the high frequency component information composed of the low frequency component information 1004, the high frequency energy information 1102, and the additional signal information 1103, and outputs the multiplexed bit stream 1005.
  • the high frequency energy information 1102 and the additional signal information 1103 are calculated in units of frames for each subband, for example. In consideration of the characteristics of the input signal 1000 in the time and frequency directions, it may be calculated in units of time obtained by further subdividing the frame in the time direction and in units of bands in which a plurality of subbands are combined in the frequency direction.
  • the high-frequency energy information 1102 and the additional signal information 1103 are calculated in time units obtained by further subdividing the frame in the time direction, the time change of the high-frequency subband signal 1001 can be expressed in more detail.
  • the total bits required to code the high frequency energy information 1102 and the additional signal information 1103 The number can be reduced.
  • the division unit in the time and frequency direction used for calculating the high-frequency energy information 1102 and the additional signal information 1103 is called a time-frequency grid, and the information is included in the high-frequency energy information 1102 and the additional signal information 1103.
  • the information included in the high frequency energy information 1102 and the additional signal information 1103 is only the high frequency energy information and the additional signal information. For this reason, the amount of information (total number of bits) is small compared to low band component information including waveform information and spectrum information of narrowband signals. I only need it. Therefore, it is suitable for low bit rate codes for wideband signals.
  • the multiplexed bit stream 1005 is separated into low-frequency component information 1007, high-frequency energy information 1105, and additional signal information 1106 by the bit stream separation unit 200.
  • the low-frequency component information 1007 is information encoded using an encoding method such as AAC, for example, and is decoded by the low-frequency component decoding unit 201 to generate a narrowband signal 1008 representing the low-frequency component.
  • Narrowband signal 1008 is divided into lowband subband signal 1009 by subband division section 202 and input to band extension section 203.
  • the low-frequency subband signal 1009 is also supplied to the subband synthesis unit 204 at the same time.
  • the band extension unit 203 reproduces the high frequency component lost due to the band limitation by copying the low frequency subband signal 1009 to the high frequency subband.
  • the high band energy information 1105 input to the band extension unit 203 includes energy information of the high band subband to be reproduced. After adjusting the energy of the low frequency subband signal 1009 using the high frequency energy information 1105, it is used as the high frequency component.
  • Band extension section 203 generates an additional signal in accordance with additional signal information included in additional signal information 1106. Here, a sine wave tone signal or a noise signal is used as the generated additional signal. The additional signal is added to the energy-adjusted high frequency component to obtain a high frequency sub-band signal 101.
  • Subband synthesizing section 204 band-synthesizes low band subband signal 1009 supplied from subband dividing section 202 and high band subband signal 1010 supplied from band extending section 203 to generate output signal 1011.
  • the gains of the copied low frequency subband signal 1009 and the additional signal are adjusted so that the energy of the high frequency subband signal 1010 becomes the energy value represented by the high frequency energy information 1105 (hereinafter referred to as target energy).
  • the high frequency subband signal 1010 is generated by adding the force to the high frequency component after energy adjustment.
  • the gain of the copied low-frequency subband signal 1009 and the additional signal can be determined, for example, by the following procedure.
  • one of the copied low-frequency subband signal 1009 and the additional signal is set as a main component of the high-frequency subband signal 1010, and the other as a subcomponent.
  • G is the main component amplitude adjustment gain
  • G is the sub component amplitude adjustment gain
  • E and N are the main component amplitude adjustment gain
  • Represent the energies of the low frequency subband signal 1009 and the additional signal, respectively. If the energy of the additional signal is normally set to 1, N l. R represents the target energy of the high frequency sub-band signal 1010, Q represents the energy ratio of the main component and the sub component, and R and Q are included in the high frequency energy information 1105 and the additional signal information 1106. Note that sqrt (') is an operator for finding the square root. On the other hand, when the additional signal is the main component and the low-frequency subband signal 1009 is the main component, the gain is determined by the following equation.
  • the low frequency subband signal 1009 and the additional signal are weighted and added to calculate the high frequency subband signal 1010.
  • an accurate energy ratio Q between the low-frequency subband signal 1009 and the noise signal to be added is added to the additional signal information 1103 generated on the code side.
  • the high frequency component code key unit 102 needs to accurately calculate the noise level of the high frequency component in the input signal.
  • Non-Patent Document 3 discloses a first conventional example of a high-frequency component code key unit 102 that calculates a noise level of a high-frequency component. 7 includes a time Z frequency dial generation unit 300, a spectrum envelope calculation unit 301, a noise level calculation unit 302, and a noise level integration unit 303.
  • the time Z frequency grid generation unit 300 uses the high frequency subband signal 1001 to Group multiple subband signals in the frequency direction and time Z frequency grid information no
  • the spectrum envelope calculation unit 301 extracts the target energy R of the high-frequency subband signal in units of time Z frequency grids, and supplies the target energy R to the bitstream multiplexing unit 103 as high-frequency energy information 1102.
  • the noise level calculation unit 302 outputs the ratio of the noise component included in the subband signal as the noise level 1101 for each subband.
  • the noise level integration unit 303 uses the average value of the noise levels in a plurality of subbands to obtain additional signal information 1103 representing the energy ratio Q in units of time Z frequency grids, and supplies the additional signal information 1103 to the bit stream multiplexing unit 103. To do.
  • X (k, 1) and Y (k, 1) denote the subband signal and the predicted subband signal of subband k, respectively.
  • a linear prediction method using a covariance method or an autocorrelation method is known.
  • the difference between the subband signal X and the predicted subband signal Y becomes small, and the value of the noise level T (k) becomes large.
  • the difference between the predicted subband signal Y and the subband signal X increases, and the value of the noise level T (k) decreases. In this way, the noise level T (k) can be calculated based on the size of the noise component contained in the subband signal.
  • the noise level integration unit 303 calculates the energy ratio Q between the low frequency subband signal and the noise signal in units of a plurality of subbands based on the time Z frequency grid information 1100. This is more than one subband unit, rather than calculating and signing the energy ratio Q for each subband unit. This is because the number of bits required for the additional signal information 1103 can be reduced by calculating the energy ratio Q. For example, N subbands from subband k to subband k + N— 1
  • the additional signal information 1103 is obtained by averaging the noise levels 1101 of N subbands from subband k to subband k + N ⁇ 1.
  • fNoise the frequency number of the additional signal information 1103, and c is a constant.
  • Patent Document 1 Japanese Translation of Special Publication 2002-536679
  • Non-Patent Document 1 "Digital Radio Musice (DRM); System Specification", E TSI, TS 101 980 VI. 1. 1, 5. 2. 6, September 2001
  • Non-Patent Document 2 AES (Audio Engineering Society) Convention Paper 55 53 ", 112th AES Convention, May 2002
  • Non-Patent Document 3 "Enhanced aacPlus general audio codec; Enhanced aacPl us encoder SBR part", 3GPP, TS 26. 404 V6. 0. 0, September 2004 Disclosure of Invention
  • the conventional additional signal information calculation method averages the noise level calculated independently for each subband, the perceptual priority of the subband is not considered. Therefore, the noise level of the sub-bands that are important to the auditory sense depends on the importance of the additional signal information.
  • the high-quality audio signal encoding device cannot be realized.
  • the method of calculating the additional signal information using the spectrum envelope has a problem that the amount of calculation increases because high resolution frequency analysis and smoothing processing are required.
  • the noise level varies greatly depending on the level of smoothness, and it is difficult to optimize the level of smoothing.
  • the present invention was invented in view of the above problems, and its purpose is to reduce the amount of additional signal information that reflects the noise level of an audibly important subband according to the degree of importance.
  • the object is to provide a technique relating to a high-quality audio signal code that can be calculated in quantity.
  • a first invention for solving the above-described problem is an input signal dividing unit for extracting a high frequency signal from an input signal, and generating a first high frequency component information by extracting a spectrum of the high frequency signal.
  • a first high-frequency component encoding unit ; a noise level calculation unit that obtains the noise level of the high-frequency signal by reflecting the importance of each frequency component; and second high-frequency component information using the noise level.
  • a second invention that solves the above-described problem is an input signal dividing unit that extracts a high-frequency signal from an input signal, and generates a first high-frequency component information by extracting a spectrum of the high-frequency signal.
  • a first high-frequency component encoding unit; a noise level calculation unit that calculates a noise level using the high-frequency signal; a correction coefficient calculation unit that calculates a correction coefficient using the high-frequency signal; and the correction A noise level correction unit that corrects the noise level using a coefficient to obtain a corrected noise level; and a second high frequency component encoding unit that generates second high frequency component information using the corrected noise level;
  • An audio encoding device comprising: a bit stream multiplexing unit that multiplexes the first high frequency component information and the second high frequency component information and outputs a multiplexed bit stream. It is.
  • a third invention for solving the above-described problem is the correction coefficient calculation according to the second invention.
  • the unit calculates a correction coefficient that reflects the importance of each frequency component of the high frequency signal.
  • the correction coefficient calculation unit calculates energy for each frequency band of the high-frequency signal, and based on the energy for each frequency band. A correction coefficient is calculated.
  • a fifth invention for solving the above-mentioned problems is characterized in that, in the second or third invention, the correction coefficient calculation unit calculates a correction coefficient having a small value at a high frequency. To do.
  • a sixth invention that solves the above-described problem is that, in the first invention, at least the noise level calculated by the noise level calculation unit reflecting the importance of each frequency component of the high-frequency signal. It is characterized by smoothing in the time direction or frequency direction.
  • the correction coefficient calculation unit calculates the frequency component according to each frequency component of the high-frequency signal.
  • the correction coefficient is smoothed at least in the time direction or the frequency direction.
  • An eighth invention for solving the above-mentioned problem is to extract a high frequency signal from an input signal, extract a spectrum of the high frequency signal to generate first high frequency component information, and The noise level is calculated by reflecting the importance of each frequency component, second high frequency component information is generated from the noise level, and the first high frequency component information and the second high frequency component information are generated. And an audio coding method characterized by outputting a multiplexed bit stream. It is.
  • a ninth invention for solving the above-described problem is to extract a high frequency signal from an input signal, extract a spectrum of the high frequency signal to generate first high frequency component information, and The noise level is obtained using the high frequency signal, the correction coefficient is obtained using the high frequency signal, the noise level is corrected using the correction coefficient to obtain the correction noise level, and the second high noise is obtained using the correction noise level.
  • An audio coding method characterized by generating band component information, multiplexing the first high band component information and the second high band component information, and outputting a multiplexed bit stream.
  • a tenth invention for solving the above-mentioned problems is the above-mentioned eighth invention, wherein the correction coefficient is When obtaining the correction coefficient, a correction coefficient is obtained corresponding to the auditory importance corresponding to each frequency component of the high frequency signal.
  • a twelfth invention for solving the above-mentioned problems is characterized in that, in the above-mentioned eighth or ninth invention, when obtaining the correction coefficient, a correction coefficient having a small value at a high frequency is calculated. To do.
  • the eighth invention at the time of obtaining the noise level, at least a noise level obtained by reflecting importance of each frequency component of the high frequency signal is used. It is characterized by smoothing in the time direction or frequency direction.
  • the correction coefficient when the correction coefficient is obtained, the correction coefficient is determined according to each frequency component of the high frequency signal.
  • the calculated correction coefficient is smoothed at least in the time direction or frequency direction.
  • a fifteenth aspect of the present invention for solving the above-described problem is a process of extracting a high frequency signal from an input signal, a process of extracting a spectrum of the high frequency signal and generating first high frequency component information, Processing for obtaining the noise level of the high frequency signal by reflecting the importance of each frequency component, processing for generating second high frequency component information using the noise level, and the first high frequency component information
  • the present invention is configured to calculate a correction coefficient corresponding to auditory importance using a high frequency sub-band signal, correct a noise level, and generate additional signal information. It is possible to accurately reflect the noise level of a sub-band that is important perceptually. For this reason, a high-quality audio encoding device can be realized.
  • the invention's effect it is possible to calculate a correction coefficient based on the perceptual importance of the input signal and correct the noise level of each subband.
  • the correction coefficient calculation of the present invention performs frequency analysis with normal resolution, the amount of computation required for high-resolution frequency analysis is reduced, and subband noise that reflects auditory importance is reflected. You can ask for the level. As a result, a high-quality audio encoding device can be realized.
  • FIG. 1 is a block diagram showing the configuration of the best mode for carrying out the first invention of the present invention.
  • FIG. 2 is an explanatory diagram showing an operation concept of a correction coefficient calculation unit in the present invention.
  • FIG. 3 is a block diagram showing a configuration of an input signal dividing unit.
  • FIG. 4 is a block diagram showing the configuration of the best mode for carrying out the second invention of the present invention.
  • FIG. 5 is a block diagram showing the configuration of the best mode for carrying out the third invention of the present invention.
  • FIG. 6 is a block diagram showing a band extension code decoding apparatus.
  • FIG. 7 is a block diagram showing a configuration of a high frequency component code key unit.
  • an audio encoding device includes an input signal dividing unit 100, a low frequency component encoding unit 101, a time Z frequency grid generating unit 300, a spectrum envelope.
  • the high frequency component code key unit 102 and the high frequency component code key unit 500 are different. Comparing these components in more detail using FIG. 1 and FIG. 7, a correction coefficient calculation unit 400 and a noise level correction unit 401 are added to the high frequency component code unit 500, and the noise level integration unit 300 is replaced by the noise level integration unit 402.
  • the correction coefficient calculation unit 400, the noise level correction unit 401, and the noise level integration unit 402 will be described.
  • the time Z frequency grid information 1100 obtained by grouping a plurality of subband signals in the time and frequency directions in the time Z frequency grid generation unit 300 using the high frequency subband signal 1001 This is transmitted to the correction coefficient calculation unit 400.
  • the correction coefficient calculation unit 400 calculates the perceptual importance of each subband using the high frequency subband signal 1001 and the time Z frequency grid information 1100, and calculates the correction coefficient 1200 of each subband to the noise level correction unit 401. Communicate to.
  • the noise level 1101 of each subband calculated by the noise level calculation unit 302 using the high frequency subband signal 1001 is also transmitted.
  • the noise level correction unit 401 corrects the noise level 1101 of each subband based on the correction coefficient 1200, and outputs the corrected noise level 1201 to the noise level integration unit 402.
  • Noise level integration section 402 calculates an average value of corrected noise levels 1103 in a plurality of subbands based on time Z frequency grid information 1100.
  • the energy ratio of the noise component is calculated in units of time Z frequency grid and output as additional signal information 1103.
  • FIG. 2 shows a part of the spectrum when the input signal 1000 is subjected to frequency analysis, where the horizontal axis represents frequency and the vertical axis represents energy.
  • N subbands from subband k to subband k + N— 1 are paired.
  • the noise component energy ratio Q in region 2 must be reflected in the additional signal information 1103 in accordance with the importance of region 2. In order to do so, it is necessary to calculate the listening and emotional importance of each subband.
  • the correction coefficient 1200 representing the perceptual importance of each subband can be calculated according to the energy of the high frequency subband signal 1001, for example.
  • Subband k to subband k
  • the correction coefficient a (k) for subband k can be expressed by the following equation, for example: .
  • E represents the energy of each subband.
  • the energy of each subband may be calculated for each time grid included in the time Z frequency grid information 1100, or may be calculated using subband signals included in a plurality of time grids.
  • the energy of the high frequency sub-band signal 1001 is used as it is, but the energy of the sub-band signal 1101 may be corrected.
  • the energy of the sub-band signal can be used for calculating the force correction coefficient in a logarithm rather than using it as it is.
  • the correction coefficient may be calculated by positively using auditory characteristics. For example, simultaneous masking that makes it impossible to perceive small sounds that are present simultaneously with loud sounds, It is also possible to calculate a correction coefficient that takes into account the effect of the continuous masking that occurs. Sounds smaller than the masking threshold cannot be perceived. Therefore, the correction coefficient corresponding to the auditory importance can be calculated by relatively reducing the correction coefficient of the subband that can be ignored for auditory perception. Conversely, the correction coefficient for subbands larger than the masking threshold may be relatively large.
  • the noise level correction unit 401 corrects the noise level 1101 of each subband calculated by the noise level calculation unit based on the correction coefficient 1200 calculated by the correction coefficient calculation unit, and the corrected noise level 1201 is input to the noise level integration unit 303. Output.
  • the corrected noise level T (k) is given by
  • T (k) a (k) X T (k)
  • the result obtained by adding a constant to the product can also be used as a corrected noise level. Furthermore, by defining the correction noise level as an arbitrary function of the correction factor 1200 and the noise level 1101.
  • the noise level integration unit 402 uses the corrected noise level 1201 to calculate the energy ratio Q of the additional signal for each frequency grid included in the time Z frequency grid information 1100 and outputs it as the attached calorie signal information 1103. For example, from subband k to subband k + N— 1
  • the input signal dividing unit 100 can be configured with a subband dividing unit 110 and a subband combining unit 111 as shown in FIG.
  • Subband dividing section 110 divides input signal 1000 into N subbands and outputs high frequency subband signal 1001.
  • the subband synthesizing unit 111 generates a low frequency signal 1002 by performing subband synthesis using M (MMN) subband signals of the low frequency of the subband signal.
  • M M
  • the down-sampling filter 112 can be used to down-sample the input signal 1000.
  • the down-sampling filter 112 includes a low-pass filter having a pass band corresponding to the band of the low-frequency signal 1002, and performs high-frequency suppression processing using the low-frequency filter before down-sampling processing. Further, as shown in FIG. 3 (c), the input signal 1000 may be output as the low frequency signal 1002 without being processed.
  • correction coefficient 1200 corresponding to auditory importance is calculated using high frequency subband signal 1001, noise level 1101 is corrected, and additional signal information 1103 is generated. Therefore, it is possible to accurately reflect the noise level of the sub-band that is important perceptually. As a result, a high-quality audio encoding device can be realized.
  • the best mode for carrying out the second invention of the present invention is that an input signal dividing unit 100, a low frequency component encoding unit 101, a time Z frequency grid generating unit 300, a spectral packet, An envelope calculation unit 301, a noise level calculation unit 302, a correction coefficient calculation unit 403, a noise level correction unit 401, a noise level integration unit 402, and a bit stream multiplexing unit 103 are included.
  • the correction coefficient calculation unit 400 is simply replaced with the correction coefficient calculation unit 403, and the other Part Are exactly the same. Therefore, the correction coefficient calculation unit 403 will be described in detail.
  • the correction coefficient calculation unit 403 calculates a correction coefficient 1202 by a predetermined method based on the time Z frequency grid information 1100 and outputs the correction coefficient 1202 to the noise level correction unit 401.
  • the correction coefficient 1202 can be calculated such that the correction coefficient 1202 takes a small value for a high frequency.
  • the correspondence relationship between the frequency and the correction coefficient 1202 can be determined to be expressed by a linear function as the simplest example, or may be determined to be expressed by a nonlinear function.
  • a high-frequency signal component is often attenuated more than a low-frequency signal component. Therefore, high-quality additional signal information 1103 can be calculated using the above-described method.
  • this embodiment uses a correction coefficient 1202 based on the characteristics of a general audio signal, the amount of calculation can be reduced as compared with the first embodiment of the present invention.
  • the third embodiment of the present invention is a computer that operates according to program 601 when the above-described first and second embodiments of the present invention are configured with program 601.
  • FIG. 5 the third embodiment of the present invention is a computer that operates according to program 601 when the above-described first and second embodiments of the present invention are configured with program 601.
  • the program 601 is read into a computer 600 (central processing unit; processor; data processing device) and controls the operation of the computer 600 (central processing unit; processor; data processing device).
  • the computer 600 central processing unit; processor; data processing unit

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Abstract

Au moyen d'un signal de sous-bande à grande portée, un coefficient de correction correspondant à l'intensité du sens auditif est calculé pour corriger un niveau de bruit et générer une information de signal additionnelle, reflétant ainsi, de manière précise, le niveau de bruit de la sous-bande intense dans le sens auditif. Ainsi, il est possible de calculer l'information de signal additionnelle reflétant le niveau de bruit de la sous-bande intense dans le sens auditif selon l'intensité par une petite quantité de calculs. La quantité de calculs peut être encore réduite au moyen d'un coefficient de correction basé sur la caractéristique d'un signal audio ordinaire.
PCT/JP2006/300112 2005-01-11 2006-01-06 Dispositif de codage audio, methode de codage audio et programme de codage audio WO2006075563A1 (fr)

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EP06702057.8A EP1840874B1 (fr) 2005-01-11 2006-01-06 Dispositif de codage audio, methode de codage audio et programme de codage audio
JP2006552903A JP5224017B2 (ja) 2005-01-11 2006-01-06 オーディオ符号化装置、オーディオ符号化方法およびオーディオ符号化プログラム

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WO2011086924A1 (fr) * 2010-01-14 2011-07-21 パナソニック株式会社 Appareil de codage audio et procédé de codage audio
WO2012050023A1 (fr) * 2010-10-15 2012-04-19 ソニー株式会社 Dispositif et procédé de codage, dispositif et procédé de décodage, et programme
JP2015228044A (ja) * 2007-05-08 2015-12-17 サムスン エレクトロニクス カンパニー リミテッド オーディオ信号の符号化及び復号化方法並びにその装置
JP2016006540A (ja) * 2009-10-07 2016-01-14 ソニー株式会社 復号装置および方法、並びにプログラム
US9583112B2 (en) 2010-04-13 2017-02-28 Sony Corporation Signal processing apparatus and signal processing method, encoder and encoding method, decoder and decoding method, and program
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